A similar function was defined in rtc_base/openssl_adapter. Moving it from net/dcsctp/common/ to rtc_base/strings/. I'm planning to use StrJoin in a video codec test (a follow-up change).
Bug: webrtc:14852
Change-Id: Ie657c03e7f9fb52c189c127af6f66ec505b512ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327322
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41166}
Before this CL, a message was identified by the triple (stream_id,
is_unordered, MID) (and yes, the MID is always present in the send
queue, even when interleaved message is not enabled.). So when a chunk
was abandoned due to e.g. having reached the retransmission limit, all
other chunks for that message in the retransmission queue, and all
unsent chunks in the send queue were discarded as well.
This works well, except for the fact that resetting a stream will result
in the MID being set to zero again, which can result in two different
messages having the same identifying triple. And due to the
implementation, both messages would get abandoned.
In WebRTC, an entire data channels is either reliable or unreliable, and
for a message to be abandoned, the channel must be unreliable. So this
means that in the case of stream resets - meaning that a channel was
closed and then reopened, an abandoned message from the old (now closed)
channel would result in abandoning another message sent on the re-opened
data channel.
This CL introduces a new internal property on messages while in the
retransmission and send queue; The "outgoing message id". It's a
monotonically increasing identifier - shared among all streams - that is
never reset to zero in the event of a stream reset. And now a message is
actually only identified by the outgoing message id, but often used
together with the stream identifier, as all data in the send queue is
partitioned by stream. This identifier is 32 bits wide, allowing at most
four billion messages to be in-flight, which is not a limitation, as the
TSN is also 32 bits wide.
Bug: webrtc:14600
Change-Id: I33c23fb0e4bde95327b15d1999e76aa43f5fa7db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322603
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40881}
This tests all of the known integer unwrappers in WebRTC
* rtc::TimestampWrapAroundHandler
* webrtc::Unwrapper<T>
* webrtc::SequenceNumberUnwrapper<T>
* dcsctp::UnwrappedSequenceNumber
The conformance tests show differences between in behavior between the
unwrappers.
This change also adds const accessors for the internal values of dcsctp::UnwrappedSequenceNumber.
Bug: webrtc:13982
Change-Id: Ia1b483d7ef5ceb43fa4d93013a76cd0251a58c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38900}
It's useful for other parts of WebRTC and there is no real reason why
it should be located in net/dcsctp.
Bug: None
Change-Id: Iccaed4e943e21ddaea8603182d693114b2da9f6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232606
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35055}
dcSCTP library users can set their custom
g_handover_state_transformer_for_test that can serialize and
deserialize the state. All dcSCTP handover tests call
g_handover_state_transformer_for_test. If some part of the state is
serialized incorrectly or is forgotten, high chance that it will
fail the tests.
Bug: webrtc:13154
Change-Id: I251a099be04dda7611e9df868d36e3a76dc7d1e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232325
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35035}
This will be useful in future socket handover tests.
Bug: webrtc:13154
Change-Id: Ia789ae971edd9d2832be088f2f8f7dd50c9ce52d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231222
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34931}
Those were found when trying to build within Chromium's codebase.
Bug: webrtc:12614
Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33896}
This is merely a container of components that have their lifetime
bound to when the socket is connected. If the socket gets disconnected
or restarted, this object (and everything it holds) will be released.
Bug: webrtc:12614
Change-Id: Ibd75760b7bf7efe9c26c4eb7cee62de8bba5410c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214340
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33869}
When the client asks for a message to be sent, it's put in the
SendQueue, which is available even when the socket is not yet connected.
When the socket is connected, those messages will be sent on the wire,
possibly fragmented if the message is large enough to not fit inside a
single packet. When the message has been fully sent, it's removed from
the send queue (but it will be in the RetransmissionQueue - which is
added in a follow-up change, until the message has been ACKed).
The Send Queue is a FIFO queue in this iteration, and in SCTP, that's
called a "First Come, First Served" queue, or FCFS. In follow-up work,
the queue and the actual scheduling algorithm which decides which
message that is sent, when there are messages in multiple streams, will
likely be decoupled. But in this iteration, they're in the same class.
Bug: webrtc:12614
Change-Id: Iec1183e625499a21e402e4f2a5ebcf989bc5c3ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214044
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33798}
Adding strong types went a little too far as these two types represent
the same sequence number. A "request sequence number" is a number, that
- when responded to - will be used as "response sequence number".
Having them separate added confusion and just a lot of type-casting.
Bug: webrtc:12614
Change-Id: I4636ea8f2252023a2d5a9b7033763e1978b1812e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214130
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33789}
Due to a previous refactoring, the SCTP packet header is only added when
the first chunk is written. This wasn't reflected in the
`bytes_remaining`, which made it add more than could fit within the MTU.
Additionally, the maximum packet size must be even divisible by four as
padding will be added to chunks that are not even divisble by four (up
to three bytes of padding). So compensate for that.
Bug: webrtc:12614
Change-Id: I6b57dfbf88d1fcfcbf443038915dd180e796191a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215145
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33760}
* `AddTo` and `Difference` are made into static methods, as one may have
believed that these modified the current object previously. The
`Increment` method is kept, as it's obvious that it modifies the
current object as it doesn't have a return value, and `next_value` is
kept, as its naming (lower-case, snake) indicates that it's a simple
accessor.
* Difference will return the absolute difference. This is actually the
only reasonable choice, as the return value was unsigned and any
negative value would just wrap.
Bug: webrtc:12614
Change-Id: If14a71636e67fc612d12759dc80a9c2518c85281
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215069
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33714}
Adding fuzzers to the build made "gn gen --check" discover a lot
of dependency errors between various components of dcSCTP.
Bug: webrtc:12614
Change-Id: I0b2dd7321aec2624da417f413c727bd11b4743e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215003
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33705}
As this library will only use StrongAlias types for all its
sequence numbers, the UnwrappedSequenceNumber class should use those
types and not the primitive underlying types (e.g. uint32_t).
This makes e.g. Unwrap() return a strong type, which is preferred.
Bug: webrtc:12614
Change-Id: Icd0900c643a1988d1a3bbf49d87b4d4d1bbfbf1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213663
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33651}
Quite a large commit, but mostly trivial. It adds all the (in dcSCTP)
supported parameters, error causes and chunks as an object model, with
serializers and deserializers. They are verified with packet captures
where available, that have been captured with Wireshark against a
reference implementation.
This _could_ be split in parameter/ as one commit, error_cause/ in the
following, and chunk/ as the third, but as each chunk/parameter is
completely isolated from the other, reviewing it should be linear with
the number of chunks/parameters and having them in more commits wouldn't
change that, taken all those three commits into account.
Bug: webrtc:12614
Change-Id: Ie83c9a22cae6e3a39e35ef26fd532837a6387a08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213347
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33625}
There are numerous identifiers and sequences in SCTP, all of them being
unsigned 16 or 32-bit integers.
* Stream identifiers
* Payload Protocol Identifier (PPID)
* Stream Sequence Numbers (SSN)
* Message Identifiers (MID)
* Fragment Sequence Numbers (FSN)
* Transmission Sequence Numbers (TSN)
The first two of these are publicly exposed in the API, and the
remaining ones are never exposed to the client and are all part of SCTP
protocol.
Then there are some more not as common sequence numbers, and some
booleans. Not all will be in internal_types.h - it depends on if they
can be scoped to a specific component instead. And not all types will
likely become strong types.
The unwrapped sequence numbers have been renamed to not cause conflicts
and the current UnwrappedSequenceNumber class doesn't support wrapping
strongly typed integers as it can't reach into the type of the
underlying integer. That's something to explore later.
Bug: webrtc:12614
Change-Id: I4e0016be26d5d4826783d6e0962044f56cbfa97d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213422
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33620}
These are quite generic utilities that are used by multiple modules
within dcSCTP. Some would be good to have in rtc_base and are simple
replicas of utilities available in abseil.
Bug: webrtc:12614
Change-Id: I9914286ced7317a34628a71697da9149d6d19d38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213190
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33609}
